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2009-11-03Merged revisions 227361 via svnmerge from lmadsen1-12/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227362 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227162 via svnmerge from lmadsen1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@227163 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Merged revisions 226384 via svnmerge from lmadsen1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226387 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225033 via svnmerge from dvossel2-0/+18
https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225310 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223756 via svnmerge from dvossel1-19/+36
https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@223759 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-05Recorded merge of revisions 222110 via svnmerge from kpfleming2-10/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221592 via svnmerge from kpfleming1-5/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221598 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221432 via svnmerge from mnicholson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221486 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221368 via svnmerge from mnick1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | 23 lines Merged revisions 221153,221157,221303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221266 via svnmerge from twilson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-16Merged revisions 219061 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219064 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-14Merged revisions 218361 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218362 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07fix documentation so it agrees with codeoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216656 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216438 via svnmerge from oej1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216645 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215955 via svnmerge from dvossel1-0/+56
https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216003 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21Merged revisions 213494 via svnmerge from qwell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r213494 | qwell | 2009-08-21 11:04:21 -0500 (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | 5 lines Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@213495 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Merged revisions 209132 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@209133 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-18Merged revisions 145293,158010 fromrmudgett1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@207286 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-17Merged revisions 207095 via svnmerge from jpeeler1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@207097 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-16Merged revisions 206873 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@206876 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203699 via svnmerge from file1-4/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203701 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-15Merged revisions 165180,200689 via svnmerge from kpfleming1-12/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@200724 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-02Merged revisions 198791 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198792 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28issue #15155 and issue #15156 from trunkghenry1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197439 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Merged revisions 197089 via svnmerge from seanbright5-9/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Change register format example to match wording.lmadsen1-1/+1
This change does not affect any other 1.6 branches as they have already been updated for other changes, which uses the word 'domain' as I have here. (closes issue #15204) Reported by: okrief git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197088 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Merged revisions 196416 via svnmerge from dvossel1-4/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196454 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-15Merged revisions 194765 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r194765 | russell | 2009-05-15 13:43:42 -0500 (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines Fix some spelling fail. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@194766 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-08Merged revisions 193194 via svnmerge from kpfleming1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines Merged revisions 193193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@193195 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-04Merged revisions 191955 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines Ensure that by default only one console channel driver is loaded This configuration file was changed to ensure that only one console channel driver (chan_oss) is loaded by default, but the change would only work if chan_console was not built. Now it will work as expected; if chan_alsa or chan_console are built and installed, they will not be loaded unless explicity requested. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@191956 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-29Revert revision 190576 after out of band discussion with transnexus.russell1-13/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@190986 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-271. Fixed the issue caused by network ID.transnexus1-0/+13
2. Fixed the issue caused by without certificate files. 3. Fixed the issue caused by number portability parameters in user part of RURI. 4. Updated for OSP Toolkit 3.5. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@190576 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186444 via svnmerge from tilghman1-0/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@186446 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186175 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@186176 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186060 via svnmerge from tilghman1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@186061 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185123 via svnmerge from rmudgett1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@185125 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24Merged revisions 183914 via svnmerge from tilghman1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@183915 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181499 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181509 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180383 via svnmerge from mmichelson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180404 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 180007 via svnmerge from mmichelson1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180008 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 178986 via svnmerge from murf1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178987 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178446 via svnmerge from tilghman1-0/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178447 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Eliminate mention of a variable which is only available in trunk.tilghman1-6/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176098 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173776 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173777 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Merged revisions 173104 via svnmerge from tilghman1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Merged revisions 172894 via svnmerge from lmadsen1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172896 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Merged revisions 172581 via svnmerge from twilson1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 Jan 2009) | 2 lines Remove incorret line from sample config ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Merged revisions 172580 via svnmerge from twilson1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172635 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172400 via svnmerge from rmudgett1-3/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172434 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172315 via svnmerge from tilghman1-12/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 Jan 2009) | 2 lines Better document mode=multirow, based upon a conversation with Jared. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172316 f38db490-d61c-443f-a65b-d21fe96a405b
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